You will generally not get definitive answers for sizing questions, and you haven't provided enough information anyway, however taking the disk size, I would suggest 2GB as a starting point; of course you may have trouble getting anything that small.

If you want to record all calls, and assuming each call lasts two minutes, you are going to require around an extra 100GB per day of retained storage,, plus any workspace your software needs (this assumes that both directions are recorded separately; I can't remember if MixMonitor records as L and R channels, or mixes into a single mono stream.

Please note that I deal with development model systems, not production ones, so cannot do the actual sizing, but these are some of the factors that might affect the sizing

Peak erlangs?

Channel technologies used?

Codecs used?

Is there any transcoding (particular expensive codecs like G.729 or GSM)?

Is any form of native bridging possible, in particular SIP direct media, and what is the peak erlang loading of traffic for which full direct media and native bridging are not possible? (Direct media removes the highest volume of traffic and the highest load on most systems, but is only possible if codecs match, the parties agree, and Asterisk not recording and has no interest in any DTMF.)

Do you use early media, or do you answer incoming calls before the outgoing leg answers (implies, at a minimum, that Asterisk will have to generate in band tones)?

Are you recording calls, and the codec used?

Use of conferences?

Use of queues?

Use of voice announcements?

What level of diagnostic logging, and how long do you intend to keep it?

What if any call accounting records, and how long do you want to keep them on the server?

Do you intend to use Asterisk Realtime Architecture and if so, what database?

david55 wrote:Please note that I deal with development model systems, not production ones, so cannot do the actual sizing, but these are some of the factors that might affect the sizing

Peak erlangs?

Channel technologies used?

Codecs used?

Is there any transcoding (particular expensive codecs like G.729 or GSM)?

Is any form of native bridging possible, in particular SIP direct media, and what is the peak erlang loading of traffic for which full direct media and native bridging are not possible? (Direct media removes the highest volume of traffic and the highest load on most systems, but is only possible if codecs match, the parties agree, and Asterisk not recording and has no interest in any DTMF.)

Do you use early media, or do you answer incoming calls before the outgoing leg answers (implies, at a minimum, that Asterisk will have to generate in band tones)?

Are you recording calls, and the codec used?

Use of conferences?

Use of queues?

Use of voice announcements?

What level of diagnostic logging, and how long do you intend to keep it?

What if any call accounting records, and how long do you want to keep them on the server?

Do you intend to use Asterisk Realtime Architecture and if so, what database?